1. Field of the Invention
The present invention provides an apparatus and method for performing call handover between a packet network system and a circuit network system.
2. Description of the Related Art
The circuit network system refers to a system that uses a scheme of assigning a fixed call circuit between users intending to enjoy a voice call, such as Global System for Mobile communication/Wideband Code Division Multiple Access Circuit Switch (GSM/WCDMA CS). The users are assumed to use a Circuit Switched (CS) scheme over the GSM/WCDMA system for a voice service.
The Long Term Evolution (LTE) system, a packet network system applied in the present invention, is a system that uses a Packet Switched (PS) scheme, which is the next generation mobile communication system evolved based on Universal Mobile Telecommunication Service (UMTS). That is, the LTE system provides a voice call or Voice over IP (VoIP) service using Internet Protocol Multimedia Subsystem (IMS) for a voice service. IMS provides communication through an unfixed path using an Internet Protocol (IP) packet, and a VoIP message is transmitted through an IMS session.
An Application Server (AS) of IMS anchors both a VoIP call and a CS call occurring in the LTE system network and the CS system. That is, for the CS call, a Mobile Service switching Center (MSC) acquires a routing address to an IMS network through a Customized Applications for Mobile Network Enhanced Logic (CAMEL) process, and it is delivered to a Media Gateway Control Function (MGCF) of the IMS network using an Integrated Services Digital Network User Part (ISUP) Initial Address Message (IAM) message. The IAM message delivered to MGCF is translated into a Session Initiation Protocol (SIP) INVITE Request and delivered to an Application Server (AS) via a Serving (S)-Call Session Control Server (CSCF). Upon receipt of the SIP INVITE Request, the AS stores information on the corresponding CS call, and delivers a SIP INVITE Request to a receiving side based on the received SIP INVITE Request message to connect a call. An SIP INVITE Request that a User Equipment (UE) has transmitted for a VoIP call is delivered to the AS via a Proxy (P)-CSCF and the S-CSCF. The AS, as in the case of the CS call, anchors a corresponding VoIP call, and delivers an SIP INVITE Request to the receiving side to connect a call. Therefore, both the CS call and the LTE VoIP call are controlled in the IMS network.
FIG. 1 is a diagram illustrating the situation where an LTE VoIP call is connected.
Referring to FIG. 1, a control signal 109 for a VoIP call is anchored from a UE for LTE (UE-LTE) 101b to an AS 107 via a P-CSCF 105 and an S-CSCF 106, and then connected again to a receiving side 108 via the S-CSCF 106. In this state, voice data 110 is delivered to a System Architecture Evolution (SAE) Anchor 104 via an Enhanced Node B (ENB) 102 and a User Plane Entity (UPE) 103, and then delivered to the receiving side 108 over the IP network.
FIG. 2 is a diagram illustrating the situation where a CS call is connected.
Referring to FIG. 2, a control signal 209 for a CS call is delivered from a UE for CS (UE-CS) 201a to an S-CSCF 206 via an MSC 203 and an MGCF 204 of an integrated Mobile Management Entity (MME), and the S-CSCF 206 sends this signal to an AS 207 to anchor it therein. Thereafter, the AS 207 sends a control signal to a receiving side 208, completing the call. In this state, voice data 210 is delivered to an SAE Anchor 205 via a Media GateWay (MGW) 202 of the MSC 203 and an MGW 204 of the integrated MME, and then delivered to the receiving side 208 over the IP network.
The UE can support both the GSM/WCDMA network and the LTE network in this way, but it has a restriction in simultaneously accessing the two types of networks. In other words, the UE cannot access the GSM/WCDMA network while communicating with the LTE network, and similarly, the UE cannot access the LTE network while communicating with the GSM/WCDMA network.
Meanwhile, there is 3GPP TS23.206 as a conventional technology for supporting handover between a VoIP call and a CS call. In TS23.206, a call undergoes call anchoring through a Voice Call Continuity (VCC) application which is an AS. However, it is assumed in TS23.206 that the UE can simultaneously access a CS network and a PS network. Therefore, the UE uses the following method in which when there is a request for handover, the UE forms a radio link to another system while maintaining the call of the old system, to generate a new session, and when the generation of the new session is completed, the UE releases the old session and the old radio link. Therefore, such a method cannot be applied when the UE, as assumed herein, cannot simultaneously access different networks.
Therefore, when the UE needs handover to the GSM/WCDMA CS network while receiving the VoIP service over the LTE system, there is a demand for a Radio Access Network (RAN) technique and an IMS session control technique for supporting the handover, and an apparatus for implementing the techniques.